The present invention relates to automatic mixing of audio signals and, more particularly, to automatic mixing of audio signals in which a gain of one or more of the audio signals is reduced or compressed, irrespective of whether the automatic mixing function would permit the gain to be higher.
A typical sound system includes four basic elements, namely one or more microphones, a microphone preamplifier or mixer, a power amplifier and a loudspeaker system. When the sound system is introduced into an acoustic environment, acoustic feedback is a concern.
Acoustic feedback occurs when direct and reflected sound from the loudspeakers arrives at the microphone at a volume greater than the original sound that entered the microphone. Such feedback generally occurs at a prominent frequency, creating a howling sound. Acoustic feedback may also occur even when the direct and reflected sound arrives at the microphone at a slightly lower volume. Indeed, the sound may still ring through the system by reducing slightly in level each cycle.
The conventional approach to reducing acoustic feedback is to insure that the loudest sound from the loudspeaker system arrives at the microphone lower (e.g., a 6 dB margin) than the original sound. This suggests that the gain (of the audio channel) must be set so that the sound level from the user of the microphone (the talker) is 6 dB louder than the reverberant sound from the loudspeaker system.
If there is only one talker using the sound reinforcement system, then it is relatively easy to maintain the 6 dB margin and insure that acoustic feedback does not occur. As additional sound sources (talkers) and/or microphones are added, however, it becomes a more difficult and complex problem to maintain proper margins and insure that acoustic feedback does not occur.
As microphones are added to the system, the gain of each microphone has to be reduced, for example, by 3 dB each time the number of open microphones is doubled. This is undesirable, however, as the theoretical maximum sound amplification is likewise reduced. Controlling the gain of each microphone so that only one microphone is on (open) at one time would permit higher amplification in each audio channel.
An additional problem with employing multiple microphones is the comb filtering effect. This occurs when sound from the talker arrives (i) at the same microphone via two different paths of different length, and/or (ii) at two open microphones located at different distances from the talker. The comb filter effect emphasizes sound at some frequencies and cancels sound at other frequencies (resulting in a notched or combed frequency response).
The comb filter effect may be lessened by insuring that sound from the talker's voice impinging on microphones other than his own is about 10 dB lower. This can be achieved by ensuring that the talker's microphone is about three times closer to the talker than any other microphone. Alternatively, the gain of microphones other than the talker's may be reduced by 10 dB.
Automatic audio mixing technology (so-called automixers) may be used to address both acoustic feedback and the comb filter effect. An automixer automatically mixes signals from multiple-microphones, without the need for a system operator. An automixer activates only those microphones that are needed and adjusts the system gain to maintain system stability. This often results in a significantly increased system gain without acoustic feedback.
Automixers employ an algorithm to “decide” how to adjust the mix of the audio signals from the microphones. Several decision algorithms exist, such as the fixed threshold method, ambience sensing, direction sensing, the scanning threshold method, the number-of-microphones-equals-one (NOM=1) method, and gain sharing.
An example of the fixed threshold approach is manifest in the VOX (voice operated switch). A detector in the microphone channel of the mixer switches the channel ON when an audio signal is present, and switches the channel OFF when the audio signal is not present. To turn the channel ON, the audio signal must be greater than a preset threshold for that channel.
The adaptive threshold approach dictates that the automixer automatically adjust its threshold level to the conditions of the space in which the microphones are located. For example, in a noisy room the automixer would increase the threshold level to prevent any of the microphone channels from being triggered ON by noise. Ambience sensing, direction sensing, and the scanning threshold method are all species of the adaptive threshold approach.
The ambience sensing approach employs a “dummy microphone” to sense the ambient noise of the space and automatically adjust the threshold level accordingly. The direction sensing approach determines the direction from which the sound source arrives to the microphone. The automixer only responds to signals having sufficient levels within a predetermined space in front of the microphone. The Scanning threshold approach involves scanning the level on all of the input audio channels and activating the channel with the highest level. The highest level channel remains active while another scan begins. If the level of active channel is still higher than the other input channels, then it remains on and the process repeats.
Although the threshold approaches above are useful, the system gain still must be reduced unless only one microphone is permitted to be on at a given time. The NOM approach employs an attenuator circuit that “counts” the number of microphones that are on in the system, and then attenuates the system output by a predetermined amount. For example, when two microphones are on, the NOM circuit attenuates the output by 3 dB to maintain NOM=1 and to prevent acoustic feedback.
The gain sharing approach employs voltage-controlled amplifiers (VCAs) to vary the gain of each audio channel instead of using a switch. The gain of each channel is adjusted by comparing its level to the level of a sum of all channel levels. The gain is computed so that the combined system gain of all microphones remains constant. In this system, the microphones with the strongest signal are given the highest gain and those with low level signals have their gain reduced.
All of the above automixing systems are problematic in that they do not address a very unpredictable factor, namely, the potential that the talker may suddenly shout, which would tend to overdrive the channel and cause clipping, acoustic feedback or other undesirable characteristics in the output from the loudspeaker system. Indeed, the gain sharing automixing approach, for example, provides that the aggregate gain of the system is shared among the audio channels, with the highest level channel receiving most of the gain. When the talker suddenly shouts, the gain sharing automixing approach dictates that the channel should continue to receive most of the gain. This does nothing to counter the fact that the excessive sound level may overdrive the system.
In accordance with the foregoing, there is a need in the art for new methods and apparatus for automatic mixing of audio signals in which a gain of one or more of the audio signals is reduced, irrespective of the automatic mixing algorithm.